On Fri, Aug 17, 2012 at 4:07 PM, Andrew Latham <lath...@gmail.com> wrote: > On Fri, Aug 17, 2012 at 4:01 PM, Joshua Colp <jc...@digium.com> wrote: >> ----- Original Message ----- >>> On Fri, Aug 17, 2012 at 2:45 PM, Juan Castro <jcas...@instant.com.br> >>> wrote: >>> > I see no indication of how to do this in sip.conf, and when I start >>> > Asterisk, it doesn't wait on port 80. >>> > >>> >>> Websocket support is being actively worked on. HTTP support should >>> be >>> enabled in manager.conf and http.conf first. >> >> Hola! >> >> The above will get the HTTP server portion going, but here's some other >> items: >> >> 1. transport=ws must be added to the peer/friend/user in sip.conf >> 2. avpf=yes must be set for that peer/friend/user as well. >> >> Depending on what you are testing with this can get you a little further. >> >> If you are using Chrome things will not quite work, yet. While they have >> made considerable progress becoming compliant with the ICE specification >> (SDP is now almost proper) it seems as though their STUN implementation is >> still not there yet. Completely valid packets sent by the library we use >> just seem to be ignored. >> >> Patience is a virtue really as things are still evolving. >> >> As well I will be working on a wiki page that will describe this stuff in >> detail. I was holding off until things were a bit more "there" but as people >> are at least trying it shall appear soon. >> >> Cheers, >> >> -- >> Joshua Colp >> Digium, Inc. | Software Developer >> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA >> Check us out at: www.digium.com & www.asterisk.org >> >> -- >> ___ > > Joshua > > Can you copy and past into a wiki page for everyone's benefit? Maybe > https://wiki.asterisk.org/wiki/display/~jcolp/WebRTC_Demo_Setup or > like page would be good. > > -- > ~ Andrew "lathama" Latham lath...@gmail.com http://lathama.net ~
s/past/paste/ oops -- ~ Andrew "lathama" Latham lath...@gmail.com http://lathama.net ~ -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
